The statements in this section merely provide background information related to the present disclosure and may not constitute prior art.
Real-Time Transport Protocol (RTP) is a protocol on Internet multi-media stream transmission, which was published by the Internet Engineering Task Force (IETF) as RFC1889. RTP is defined as working under the transmission conditions like one-to-one or one-to-many. The target of RTP is to provide time information and to accomplish stream synchronization. The service of RTP is mainly to provide load type identification, sequence numbering, timing location and transmission monitoring. The typical application of RTP may be established on User Datagram Protocol (UDP), Transmission Control Protocol (TCP), Asynchronous Transfer Mode (ATM) or other protocols. RTP usually could provide real time data transmission, but could not provide reliable transmission for sequential data packets, or stream control or congestion control. Usually RTP relies on RTP Control Protocol (RTCP) to provide these services.
RTCP provides administration on transmission quality, exchanging control information among present application processes. During a session of RTP, every participant delivers RTCP packet periodically, in the packet are statistic information of the sent packets and the lost packets, so that a server can dynamically change transmission rate or even payload type according to these information. When RTP and RTCP are applied cooperatively, transmission efficiency could be improved with efficient feedback and fewer overhead, which is applicable to real-time data transmission on the internet.
QoS may be defined as follows: Quality of service is the ability to provide different priority to different applications, users, or data flows, or to guarantee a certain level of performance to a data flow. For example, a required bit rate, delay, jitter, packet dropping probability and/or bit error rate may be guaranteed. Quality of service guarantees are important especially for real-time streaming multimedia applications such as voice over IP, online games and IP-TV, since these often require fixed bit rate and are delay sensitive, and also in networks where the capacity is a limited resource, for example in cellular data communication network.
An operator often develops streaming media QoS service according to a certain convergence ratio. In order to provide variant service, the level of users is often differentiated. A user with a higher level could get better QoS. A user with a lower level could get relevant QoS when the network is vacant or idle or with enough bandwidth. But when the network is congested, the packet of the user with the lower level may be discarded earlier than the packet of the user with the higher level; the quality of lower level service obtained then would be awful. When the bandwidth of carrier is heavier than the whole available bandwidth, and the priorities of users are the same in the mean time, all user flows might be discarded. Thus all users would encounter a delay of information, for example the showing of mosaic may come out.